[..] |
ASTERISK-29000: internationalization: UTF-8 character in channel variables causes crashes |
ASTERISK-29001: chan_pjsip does not process or forward 181 responses |
ASTERISK-29002: asterisk 17.x crash |
ASTERISK-29003: Asterisk 13 PJSIP: Irregular Response to OPTION-Requests |
ASTERISK-29004: SIP/2.0 488 Not Acceptable Here when configured with PJSIP/TLS |
ASTERISK-29005: core: Safe sleep can cause "Exceptionally long queue length queuing" and Asterisk freeze |
ASTERISK-29006: Asterisk 16 on Centos 7 with 16 CPUs and 64 GB RAM cannot process 5CPS |
ASTERISK-29007: cdr_mysql: constructs invalid SQL queries on reconnect |
ASTERISK-29008: Adding an XMPP component user does not approve subscription |
ASTERISK-29009: app_amd: Detection issues when silence is not transmitted |
ASTERISK-29010: Allow disabling of FollowMe prompt |
ASTERISK-29011: chan_sip: ToHost property not cleared on reload |
ASTERISK-29012: cel_odbc skips any tables it cannot connect to at config time |
ASTERISK-29013: res_pjsip: Asterisk doesn't stop sending invites (with auth) on 407 replies |
ASTERISK-29014: res_pjsip_session: Re-INVITE collisions aren't handled correctly |
ASTERISK-29015: res_pjsip_pubsub: ps_subscription_persistence Alembic does not include generator_data |
ASTERISK-29016: Custom PJSIP Header in REFER |
ASTERISK-29017: pjsip: As of 2.9 with newer OpenSSL "tlsv1" method is TLSv1.3 only |
ASTERISK-29018: pbx: Channel redirect might be missed if redirected channel is executing dialplan |
ASTERISK-29019: cel: Asterisk crashing with switch usage (realtime switch) |
ASTERISK-29020: Regression: Fix for app_queue: track masquerades in app_queue to avoid leaked stasis subscriptions broke queue log |
ASTERISK-29021: [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified versions |
ASTERISK-29022: Crash when manipulating PJSIP invite dlg ref counts |
ASTERISK-29023: Include PJSIP's check_contact option |
ASTERISK-29024: pjsip: Route Header in Cancel request incorrectly set |
ASTERISK-29025: pbx_spool: Call remains up when it shouldn't be |
ASTERISK-29026: app_voicemail: When a voicemail is marked as "Urgent", it is not sent by email/processed by the mailcmd command |
ASTERISK-29027: Implement support for History-Info |
ASTERISK-29028: Queue not ring PJSIP extension |
ASTERISK-29029: Voicemail "pollmailboxes"-option not working, bug in function handle_subscribe |
ASTERISK-29030: res_rtp_asterisk: Additional RTP-frame (with wrong SSRC) gets inserted when switching from progress to established |
ASTERISK-29031: Deadlock and stop making calls |
ASTERISK-29032: DMTF interoperability |
ASTERISK-29033: res_pjsip_session: Aggressively terminates session on failed re-INVITE |
ASTERISK-29034: Lastpause of realtime members is reseting |
ASTERISK-29035: chan_local: Multistream support breaks T.38 faxing |
ASTERISK-29036: Regression: MWI polling no longer works |
ASTERISK-29037: segfault in opus codec stats transmission |
ASTERISK-29038: res_pjsip_registrar: Request to remove Contact fails |
ASTERISK-29039: chan_mobile: Crash on dial/answer |
ASTERISK-29040: res_speech: Assertion on format |
ASTERISK-29041: manager: Logging in results in connection closure |
ASTERISK-29042: res_parking: Parker UUID is no longer copied |
ASTERISK-29043: app_queue: Leave empty sometimes not recorded as abandoned |
ASTERISK-29044: call_pickup: Crash during call_pickup test |
ASTERISK-29045: app_queue: Does not assign the first call in queue when one or more members are not in the range of MIN and MAX penalty |
ASTERISK-29046: pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension |
ASTERISK-29047: dpma: Crash in body generator |
ASTERISK-29048: chan_iax2: "iax2 show registry" shows host for perceived |
ASTERISK-29049: Memory Leak caused by fix for ASTERISK-28445 |
ASTERISK-29050: Make call from softphone outbound cell phone |
ASTERISK-29051: res_pjsip_sdp_rtp: Does not set correct values on RTP instance when "auto" DTMF is used |
ASTERISK-29052: res_pjsip_session: Deferred re-INVITE without SDP results in a=sendonly |
ASTERISK-29053: echec de la connection sans fil des clientsips au server asterisk 16 apres avoir bien configurer les fichiers sip.conf |
ASTERISK-29054: Logger: Add debug logging categories |
ASTERISK-29055: Create a Bridge with video_single mode |
ASTERISK-29056: Increase reg_server column size for ps_contacts table realtime |
ASTERISK-29057: pjsip: Crash on call rejection during high load |
ASTERISK-29058: app_bridgewait: On timeout returning back to wrong context after redirects |
ASTERISK-29059: Asterisk sends endless INVITE requests even call is ended |
ASTERISK-29060: app_voicemail: No error/warning is provided if mailcmd does not exist |
ASTERISK-29061: chan_dahdi: Recurring log message with "mwimonitor=fsk" |
ASTERISK-29062: Real-time Music On Hold Not Playing After Commit 14733 |
ASTERISK-29063: PJSIP leaves endpoint in Unavail state when successful qualification happens right after Unavail state change |
ASTERISK-29064: res_rtp_asterisk: Audio Delay on WebRTC Call from Delayed ICE Completion |
ASTERISK-29065: Register SIP Softphone - Not conneted |
ASTERISK-29066: Duplicate Hold event |
ASTERISK-29067: Asterisk FastAGI Record Empty File |
ASTERISK-29068: Crashes due to "channel.c: Resolve issue with receiving SIP INFO packets for DTMF" |
ASTERISK-29069: chan_sip: FRACK on incoming handling of REGISTER |
ASTERISK-29070: CLONE - Any curl response checks out as valid even if 404 is returned. |
ASTERISK-29071: app_confbridge: Memory rises when jitterbuffer enabled and muting over AMI occurs |
ASTERISK-29072: Manager EventFilter(Blacklist) not working |
ASTERISK-29073: SNOM3xx series FW 8.7.5.35 get error 404 on incoming call |
ASTERISK-29074: SIP-Reregister timeout based on absolute time |
ASTERISK-29075: CLONE - Any curl response checks out as valid even if 404 is returned. |
ASTERISK-29076: Suggestion - Add the domain supported registration |
ASTERISK-29077: I have the whole phone system down |
ASTERISK-29078: send dtmf to an outgoing call to the pstn |
ASTERISK-29079: Issue Syncing PJSIP Endpoint Statuses across multiple asterisk Instances/PBX |
ASTERISK-29080: SIP calls disconnected automatically |
ASTERISK-29081: res_stasis: Add compare function for bridges moh container |
ASTERISK-29082: hi, when i want to go in to asterisk folder (cd /etc/asterisk) i received permission denied |
ASTERISK-29083: Do not build chan_sip by default as it is now deprecated |
ASTERISK-29084: TLS registration crashing asterisk |
ASTERISK-29085: func_curl: Segmentation fault when using CURL after setting httpheader CURLOPT |
ASTERISK-29086: Option "send_pai=yes" is not working? |
ASTERISK-29087: Not getting extension or destinations channel in queue abondon cdr |
ASTERISK-29088: Autodestruct on dialog. Asterisk stopped working. |
ASTERISK-29089: RTP Ports not cleared after hangup |
ASTERISK-29090: Asterisk tries to persist TCP and TLS Subscriptions and always fails |
ASTERISK-29091: Crash when ast_translator_build_path fails |
ASTERISK-29092: Asterisk crashes because of segfault |
ASTERISK-29093: Asterisk crash on 16.13.0 in media file playback |
ASTERISK-29094: asterisk integration |
ASTERISK-29095: 401 Unauthorized on every request |
ASTERISK-29096: chan_sip: Use of SIP_CODEC results in incorrect formats |
ASTERISK-29097: res_pjsip_config_wizard: Crash when freeing string when failing to add extension |
ASTERISK-29098: pjsip: Crash in pjnath on cache timeout |
ASTERISK-29099: res_musiconhold: Realtime MOH only loads a single entry |
ASTERISK-29100: chan_sip: Crash on high latency link |
ASTERISK-29101: chan_sip: Deadlock with res_rtp_asterisk |
ASTERISK-29102: What version of Asterisk firmware does Switchvox 7.5.1 use? |
ASTERISK-29103: Downloaded installer doesn't work |
ASTERISK-29104: func_jitterbuffer: Calling JITTERBUFFER multiple times does not work properly |
ASTERISK-29105: chan_pjsip: 180 Ringing with SDP not changed into progress |
ASTERISK-29106: Survey after Hangup |
ASTERISK-29107: IP address in SDP written as hex |
ASTERISK-29108: resource_endpoints.c : Memory leak if endpoint not found |
ASTERISK-29109: res_pjsip_session: Asterisk 18 does not progress calls due to codec negotiation after upgrading from Asterisk 16 |
ASTERISK-29110: res_pjsip_sdp_rtp: Asterisk does not increment session version information in late SDP reinvite scenario |
ASTERISK-29111: res_pjsip_outbound_registration: Documentation for TCP/TLS transports. |
ASTERISK-29112: Invalid UTF-8 string (problem with umlauts in callerid) |
ASTERISK-29113: we want to re direct all local calls to Huwaei soft switch server so that all records of call duration should be recorded at the Huwaei soft switch server . now we what we are looking is how to go about it in the asterisks dial plan |
ASTERISK-29114: we want to re direct all local calls to Huawei soft switch server so that all records of call duration should be recorded at the Huawei soft switch server . now we what we are looking is how to go about it in the asterisks dial plan |
ASTERISK-29115: Asterisk 17.7 PJSIP Pjpool Release Crash |
ASTERISK-29116: Asterisk hang in pbx.c on thread exit |
ASTERISK-29117: Zero wrapuptime is NOT respected in a ringall queue and no ringinuse |
ASTERISK-29118: VoiceMail() should have an option to play greetings as Early Media |
ASTERISK-29119: res_odbc: Hang in libodbc when connecting |
ASTERISK-29120: Crash: ast_bridge_channel_queue_frame on bridge softmix |
ASTERISK-29122: Unable pjsip direct_media between endpoint for vp8 codec |
ASTERISK-29123: logger.conf.sample missing comment mark on line 115 |
ASTERISK-29124: res_pjsip: flow transport broken for outbound requests |
ASTERISK-29125: we want to re direct all local calls to Huawei soft switch server so that all records of call duration should be recorded at the Huawei soft switch server . now we what we are looking is how to go about it in the asterisks dial plan |
ASTERISK-29126: How to do Cluster in 2 server |
ASTERISK-29127: SIGABRT Asterisk crash occurred with Asterisk 13.36 |
ASTERISK-29128: res_srtp: Authentication failure after hold/unhold |
ASTERISK-29129: pbx_lua: Dial subroutine leads to audio issues and FRACKs |
ASTERISK-29130: prometheus: Crash when scraping bridge |
ASTERISK-29131: Was using autodomain in sip.conf, now in pjsip how do I direct calls to a context by domain name when they come from the same location.. |
ASTERISK-29132: Problema com o Read |
ASTERISK-29133: Problem setting up ssl connection: error:00000001:lib(0):func(0):reason(1), Internal SSL error |
ASTERISK-29134: app_mixmonitor: Crash in AMI StopMixMonitor |
ASTERISK-29135: How to translate the computerized unwanted audio callee of on going call which is not configured in caller |
ASTERISK-29136: config: Sample features.conf incorrectly includes " around sound files |
ASTERISK-29137: pjsip: Broken pipe when sending requests |
ASTERISK-29138: Asterisk doesn't process ACK on high CPS |
ASTERISK-29139: Asterisk 17 Pjsip dynamic realtime only allow registration via sippeers table |
ASTERISK-29140: ARI: Can't get BridgeCreated events without subscribing to ALL events |
ASTERISK-29141: IVRS Dialplan |
ASTERISK-29142: sip_to_pjsip.py: doesn't read globbed includes |
ASTERISK-29143: res_http_media_cache: HTTP media cache stored hardcoded in /tmp |
ASTERISK-29144: GCC Warnings with OPTIMIZE=-Og make |
ASTERISK-29145: GCC Warnings with OPTIMIZE=-Os make |
ASTERISK-29146: GCC Warnings: ‘%s’ directive argument is null. |
ASTERISK-29147: Chan PJSIP + WebRTC + Chan SIP Setup. |
ASTERISK-29148: AST_MODULE_INFO no, MODULEINFO depend |
ASTERISK-29149: res_pjsip may crash on load_module (two times) |
ASTERISK-29150: Block outgoing calls (Maybe Very fast ticket) |
ASTERISK-29152: pthread_kill(): asterisk killed by SIGSEGV |
ASTERISK-29153: __pthread_mutex_lock(): asterisk killed by SIGSEGV |
ASTERISK-29154: SIGABRT in ast_channel_snapshot_create |
ASTERISK-29155: app_queue: Deadlock between queues container and individual queues |
ASTERISK-29156: 13.37.1 doesn't build codec_opus.so |
ASTERISK-29157: audit: Improve log messages |
ASTERISK-29158: res_rtp_asterisk: Ability to disable RTCP AMI messages |
ASTERISK-29159: ari: Add support for configuring variables/dialplan functions to exist in StasisEnd |
ASTERISK-29160: app_mixmonitor: Ability to pause/unpause, and AMI events |
ASTERISK-29161: Incorrect setup of recall channels |
ASTERISK-29162: dateformat ignored for CEL ODBC connection |
ASTERISK-29163: STRPTIME does not work at all |
ASTERISK-29164: ast_write: Failed to write data to channel monitor write stream |
ASTERISK-29165: res_pjsip: malformed header Accept-Encoding in OPTIONS response |
ASTERISK-29166: CDR_PROP, setting party_a, and expectations |
ASTERISK-29167: T.38 fax relay won't pass through local channels in recent versions |
ASTERISK-29168: Asterisk crashes during call transfer |
ASTERISK-29169: res_stir_shaken: Wrong CID used when looking up certificates |
ASTERISK-29170: Conflict hints with extenpatternmatchnew=true |
ASTERISK-29171: codec list in INVITE is reversed when using remote |
ASTERISK-29172: codec list in 200/183 is not using the common denominator |
ASTERISK-29173: Media cache URL requests allow infinite redirects |
ASTERISK-29175: res_pjsip_stir_shaken: Fix module description |
ASTERISK-29176: Private REDIRECTING party ID data not used in PJSIP headers |
ASTERISK-29177: Sharing information and files |
ASTERISK-29178: Asterisk-16 version: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
ASTERISK-29179: Asterisk-16 version: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
ASTERISK-29180: sip_to_pjsip.py: Default rewrite_contact=yes might be missing. |
ASTERISK-29181: while installing asterisk over Centos7 i am facing issue when i execute "make install" |
ASTERISK-29182: Audio/Video Codecs: Alternative formats get lost. |
ASTERISK-29183: Audio/Video Codecs: Format parameters get lost. |
ASTERISK-29184: memory leak manager.c:purge_old_stuff not scheduled when manager.conf enabled=no |
ASTERISK-29185: chan_pjsip: Endpoint: allow = all is broken. |
ASTERISK-29186: chan_pjsip: Endpoint not registered: log level 6: assert! |
ASTERISK-29187: stasis.c: FRACK!, Failed assertion bad magic number 0x0 for object in publish_msg |
ASTERISK-29188: null media causing the Asterisk crash |
ASTERISK-29189: SIP: TCP/TLS server: Uses SIP Contact not IP:port (on default). |
ASTERISK-29190: chan_sip / pjsip: Ability to use ephemeral TCP/TLS port in signalling can improve interop |
ASTERISK-29191: tel: URI in Diversion header causes crash |
ASTERISK-29192: Asterisk service is crashing in the pjsip transport functions due to callback are NULL. |
ASTERISK-29193: Channel Lockup when Hangup() is used in 'h' extension |
ASTERISK-29194: PJSIP NAT - rtp_symmetric not working |
ASTERISK-29195: app_queue: AMI QueueMemberStatus event property "InCall" not updated |
ASTERISK-29196: res_pjsip: Segmentation fault |
ASTERISK-29197: Crash in rewrite_route_set when Record-Route header is set |
ASTERISK-29198: allow=audio,video,text: First codec must be an audio codec. |
ASTERISK-29199: app_queue: Calls not connected |
ASTERISK-29200: Regression for preserving 'Contact' on a transaction. |
ASTERISK-29201: Crash occurs when Transfer and execute Hangup before the Transfer result |
ASTERISK-29202: res_pjsip: Cannot send OPTIONS to endpoint when transport= is unset and multiple explicitly bound transports |
ASTERISK-29203: res_pjsip_t38: Crash when changing state |
ASTERISK-29204: app_directory: Doesn't immediately accept selection |
ASTERISK-29205: res_rtp_asterisk: Asterisk crashes when making hold/unhold from webrtc client |
ASTERISK-29206: T.38 does not work through Local Channels |
ASTERISK-29207: Loop custom_beep when |
ASTERISK-29208: res_pjsip: AMI Action PJSIPShowContacts returns No Contacts found |
ASTERISK-29209: Debug messages printed by scope trace might be missing newlines |
ASTERISK-29210: res_pjsip: Crash when examining transport |
ASTERISK-29211: res_musiconhold: Segfault on realtime music on hold without entries |
ASTERISK-29212: asterisk stop automatic |
ASTERISK-29213: call sound through motherboard sound catd for E1 PRI calls |
ASTERISK-29214: ARI: specifying "CALLERID(subaddr)" variables on channel create doesn't work |
ASTERISK-29215: res_pjsip_session: NULL active_media_state topology caused asterisk crash |
ASTERISK-29216: contrib: systemd asterisk service for centos8 or other newer linux versions |
ASTERISK-29217: LOCK() can grant the same lock to multiple channels spuriously |
ASTERISK-29218: res_pjsip: segfault during UDP registration when flow transport is configured |
ASTERISK-29219: res_pjsip_diversion: Crash if Tel URI contains History-Info |
ASTERISK-29220: After T38 reinvite response of 488 a subsequent G711 reinvite is not processed correctly. Instead the previous T38 session media is used |
ASTERISK-29221: res_ari_bridges: improper error code when trying to delete a non-stasis bridge |
ASTERISK-29222: chan_sip: Hold/Resume an sRTP call on a video enabled user-agent. |
ASTERISK-29223: MOH playlist: doesn't open HTTPS URL |
ASTERISK-29224: asterisk-16 takes over an hour to clear the MWI light |
ASTERISK-29227: res_pjsip_diversion: sending multiple 181 responses causes memory corruption and crash |
ASTERISK-29228: Documentation around threadpool_max_size is inconsistent |
ASTERISK-29229: Stasis/messaging: text messages not dispatched to all subscribers when using generic subscription |
ASTERISK-29230: pjsip: Asterisk goes crazy and massively spams logfile if registration can't be send |
ASTERISK-29231: pjsip: SIGSEGV in CLI if no trunk is registered |
ASTERISK-29232: Memory Leak since 16.13.0 |
ASTERISK-29233: faxdetect_timeout does not work with channel SIP |
ASTERISK-29234: Queue Gosub option runs before channels are bridged |
ASTERISK-29235: res_pjsip_nat: Contact is rewritten on REGISTER responses with external_signaling_address |
ASTERISK-29236: ari: dial route does not use originator in Dial event for caller |
ASTERISK-29237: chan_sip: SDP: m=video is parsed even when disabled. |
ASTERISK-29238: chan_sip: SDP: Offers without any enabled stream are accepted. |
ASTERISK-29239: pbx: Using a = in a parameter for Set results in incorrect parsing |
ASTERISK-29240: chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable |
ASTERISK-29241: pjsip / register: wrong port used in Contact and Via if multiple transports are defined. |
ASTERISK-29242: pjsip: Documentation for transport handling is short / partly wrong |
ASTERISK-29243: CSeq :INVITE |
ASTERISK-29244: Add MixMonitorStart / Stop / Mute AMI events |
ASTERISK-29245: Few incoming calls rejected no compatible codecs, not accepting this offer |
ASTERISK-29246: chan_ooh323, dtmf transit problem |
ASTERISK-29247: Asterisk create too many threads |
ASTERISK-29248: res_pjsip_session: res sometimes uninitialized reported by compiler Clang. |
ASTERISK-29249: Bug Report #21J59 (User enumeration through the groupuserpicker api resource - CVE-2019-8449) |
ASTERISK-29250: Bug Report #21J60 (Missing access control exposing detailed information on all users admin) |
ASTERISK-29251: Bug Report #21J61 (allowing an unauthenticated attacker to enumerate whether a user exists on the Jira or not ) |
ASTERISK-29252: TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER) failure SIP code |
ASTERISK-29253: Incorrect bridging on transfer |
ASTERISK-29255: res_rtp_asterisk: Crash when using TURN support |
ASTERISK-29256: PJSIP is communicating directly with a name server |
ASTERISK-29257: "Received a REFER without a parseable Refer-To" from a technicolor gateway |
ASTERISK-29258: chan_sip: Audio stream rejected, Other stream present: Invalid SDP. |
ASTERISK-29259: channel: Allow text+video media streams, again. |
ASTERISK-29260: sRTP Replay Protection ignored; even tears down long calls |
ASTERISK-29261: res_pjsip: user=phone validation fail for isup numbers containing *# |
ASTERISK-29262: Support of various URL-schemes by MoH |
ASTERISK-29263: Device state not being evaluated correctly with devicestate_busy_at option enabled |
ASTERISK-29264: Certain IAX2 authentication failures cause Asterisk to crash |
ASTERISK-29265: chan_sip: Allow text+video media streams, again. |
ASTERISK-29266: ICE Role conflict with an unauthorized session |
ASTERISK-29267: H.263+ Format Attribute Module: not RFC 4629 |
ASTERISK-29268: Format Attribute Modules: Parameter Names are Case-Insensitive |
ASTERISK-29269: H.265: Format Attribute Module not implemented |
ASTERISK-29270: VP9: Format Attribute Module not implemented |
ASTERISK-29271: How to fetch existing peers/extensions from Asterisk DB by enabling Asterisk Manager Interface ? |
ASTERISK-29272: chan_iax2: Full URIs don't implement secure media checks properly |
ASTERISK-29273: Incorrect off-hold on ReINVITE via Replaces |
ASTERISK-29274: Possible Memory Leak in PJSIP TLS Transport |
ASTERISK-29275: Support of MIME-type for wav16 |
ASTERISK-29276: stun: Implementation causes delay and does not work in all network topologies. |
ASTERISK-29277: Unable to load module app_voicemail.so |
ASTERISK-29278: Forensic Analysis on Asterisk PBX 13.5.0 |
ASTERISK-29279: aasterisk rebooting unexpectedly |
ASTERISK-29280: chan_sip: Allow peers without audio (text+video). |
ASTERISK-29281: Pass-through Media Formats are added to SDP Offer. |
ASTERISK-29282: Frequently getting auto destruct messages even after restarting server or only asterisk (not entire server) |
ASTERISK-29283: 18.1.1 and greater crash on Sip over WebSocket registrations |
ASTERISK-29284: Conditinals applications are not working with glibc 2.17-322 |
ASTERISK-29285: res_format_attr_h264: level_idc can be downgraded |
ASTERISK-29286: chan_pjsip: Deadlock between sending response and transaction layer |
ASTERISK-29287: app.h: C++ compatibility broken |
ASTERISK-29288: Expired SSL Cert on forums.*.org |
ASTERISK-29289: Asterisk randomly crashes with a segmentation fault |
ASTERISK-29290: pjsip logger pcap doesn't seem to work |
ASTERISK-29291: connect signalwire to asterisk issue |
ASTERISK-29292: removing extension from pjsip does not remove it from the prometheus metrics without asterisk restart |
ASTERISK-29293: res_config_pgsql: Limit realtime_pgsql() to return one (no more) record |
ASTERISK-29294: Hi, I need Asterisk IVR Flow Development IDE . How I can get |
ASTERISK-29295: chan_pjsip: Does not respect SRCUPDATE and SRCCHANGE control frames |
ASTERISK-29296: ari: Bridge is partially invalid when it shouldn't be |
ASTERISK-29297: say: Y2021 problem – Asterisk cannot say year 2021 in Dutch |
ASTERISK-29298: Crash in WebRTC Registration on 18.2 |
ASTERISK-29299: libmyodbc in ubuntu server 20.04 using asterisk 13 |
ASTERISK-29300: res_rtp_asterisk: When native local bridging the remote SSRC becomes permanent |
ASTERISK-29301: CLONE - libmyodbc in ubuntu server 20.04 using asterisk 13 |
ASTERISK-29302: res_ari: Format cache FRACK when invoking operations |
ASTERISK-29303: pjsip: Re-invite occurs when it shouldn't |
ASTERISK-29304: Caller filtering |
ASTERISK-29305: ASTERISK-29203 / AST-2021-002 -- Another scenario is causing a crash |
ASTERISK-29306: strings: Incorrect use of __attribute__((pure)) in ast_str_to_lower definition |
ASTERISK-29307: chan_pjsip: Crash during hangup when pushing to taskprocessor |
ASTERISK-29308: logger: JSON logging can be cut off with large contents |
ASTERISK-29309: Transfered Calls CDR Issues |
ASTERISK-29310: voicemail anounce with date stops working after 2020 |
ASTERISK-29311: res_odbc_transaction sets forcecommit default value based on isolation level instead of forcecommit |
ASTERISK-29312: res_fax: asterisk fails to publish the Stasis and ReceiveFax status messages if the remote Station ID contains invalid UTF-8 characters |
ASTERISK-29313: res_pjsip_refer: Segfault in progress notify |
ASTERISK-29314: MixMonitor stops when use Transfer key of a IP Phone |
ASTERISK-29315: res_pjsip: re-registration gets stuck if setting initial auth credentials fails |
ASTERISK-29316: Asterisk service is failed due to segmentation fault about twice a week |
ASTERISK-29317: pbx_spool: Call file comments using '#' can not be escaped |
ASTERISK-29318: listen /barge concern |
ASTERISK-29319: Regression: call-id no longer printed in the console session |
ASTERISK-29320: res_pjsip_sdp_rtp: Codec preference order of remote is not correct on unhold |
ASTERISK-29321: sorcery: Add support for more intelligent reloading. |
ASTERISK-29322: Can a Asterisk instance talk to a Cisco 12.5 CUCM |
ASTERISK-29323: Asterisk error 14 |
ASTERISK-29324: app_queue: Queuerule still works in queue, even after unset it |
ASTERISK-29325: res_pjsip_registrar: Include source IP address and port in log messages |
ASTERISK-29326: asterisk: Update copyright/company |
ASTERISK-29327: Issue related to Transfer of call to external number : Constant poor audio and frequent drop of call |
ASTERISK-29328: translate.c: possible buffer overflow when upsampling |
ASTERISK-29329: app_dial: DTMF to 'D' option gets duplicated if there are multiple progress events |
ASTERISK-29330: can not dial other endpoint |
ASTERISK-29331: publish_msg: FRACK!, Failed assertion bad magic number |
ASTERISK-29332: Ring group |
ASTERISK-29333: Can't use SSH with gerrit |
ASTERISK-29334: Asterisk crashes with no message |
ASTERISK-29335: xml: Embed module information into core XML documentation. |
ASTERISK-29336: documentation: Fix inconsistent support levels |
ASTERISK-29337: menuselect: Add ability to set deprecated in and removed in versions for modules |
ASTERISK-29338: documentation: pjsip show qualify documentation inconsistent with behaviour |
ASTERISK-29339: loader: Let's output warnings for deprecated modules! |
ASTERISK-29340: stasis: PJSIP endpoint taskprocessor congestion after upgrading |
ASTERISK-29341: I have asterisk 16 I set queue timeout in app 1200 sec and in conf timeout=0 but not work more than 120 seconds at ringing before queue priority |
ASTERISK-29342: basePath in /rest-api/resources.json is always replaced with http:// scheme |
ASTERISK-29343: app_queue: Wrong order in queue_log when disconnect PJSIP call while on hold |
ASTERISK-29344: ami: Contact events getting dropped |
ASTERISK-29345: Queue_Log file empty |
ASTERISK-29346: Installation of Spitter |
ASTERISK-29347: Handling Inbound Calls (Number does not exist) |
ASTERISK-29348: menuselect doesn't return errors in many cases |
ASTERISK-29349: Silent voicemail option is not completely silent |
ASTERISK-29351: Qualify pjproject 2.12 for Asterisk |
ASTERISK-29352: res_rtp_asterisk: Fix frame delivery time when SSRC changes |
ASTERISK-29353: Qualify jansson 2.14 for asterisk |
ASTERISK-29354: res_pjsip: Allow partial reloading of transports |
ASTERISK-29355: app_queue: Queue member status message sent even if status doesn't change |
ASTERISK-29356: manager: Bridge action is rejected if either party is already a member of a bridge |
ASTERISK-29357: Inconsistent info in QueueMemberStatus |
ASTERISK-29358: chan_pjsip: Trace message for progress is output even if frame is not queued |
ASTERISK-29359: res_pjsip: Allow to define a transport without listener |
ASTERISK-29360: Asterisk, Mysql, Odbc (Unknown column 'data' in 'field list') |
ASTERISK-29361: ast_restart: Give time for NOTIFY response handling |
ASTERISK-29362: app_confbridge: AMI Event "ConfbridgeLeave" may not contain CONFBRIDGE_RESULT |
ASTERISK-29363: res_pjsip_sdp_rtp / bridge_softmix: Removal of stream on other participants does not occur |
ASTERISK-29364: res_rtp_asterisk: standard deviation miscalculation |
ASTERISK-29365: taskprocessor: Can cause assert at shutdown |
ASTERISK-29366: opus: encode testsuite test expects comma separated list of tones to work with duration |
ASTERISK-29367: app_voicemail: reload causes voicemail taskprocessor overload with ODBC and pollmailboxes |
ASTERISK-29368: Asterisk 18 Recordings REST API |
ASTERISK-29369: Asterisk 18 Recordings REST API |
ASTERISK-29370: chan_sip does not recognize application/hook-flash |
ASTERISK-29371: Asterisk exiting due to modules not being loaaded |
ASTERISK-29372: file.c switch does not account for flash events |
ASTERISK-29373: res_rtp_asterisk: Flash events are duplicated |
ASTERISK-29374: res_prometheus: Crash when scraping channels |
ASTERISK-29375: I can not make a call via zoiper (VOIP / SIP App) |
ASTERISK-29376: res_rtp_asterisk: Coredump with t.140 RED enabled |
ASTERISK-29377: cpool_release_pool "double free or corruption (out)" |
ASTERISK-29378: res_prometheus: Crash when scraping bridges and creating a bridge at the same time |
ASTERISK-29379: Segfault - ast_channel_is_multistream (chan=0x0) at channel_internal_api.c:1590 |
ASTERISK-29380: Add Flash AMI event to handle flash events |
ASTERISK-29381: chan_pjsip: Remote denial of service by an authenticated user |
ASTERISK-29384: svn error |
ASTERISK-29385: Asterisk sometimes freezes |
ASTERISK-29386: Oh dear… we couldn’t allocate a port for RTP instance |
ASTERISK-29387: res_srtp fails to load on Fedora 34 |
ASTERISK-29388: Website has broken link |
ASTERISK-29389: Add PJSIP_HEADERS() and ability to read header by pattern |
ASTERISK-29390: Create pjsip testsuite test for receiving multiple auth challenges |
ASTERISK-29391: VoiceMail does not cancel recording on rerecord hangup |
ASTERISK-29392: chan_iax2: Asterisk crashes when queueing video with format |
ASTERISK-29393: app_mixmonitor: B option can cause fallthrough errors (and not work) |
ASTERISK-29394: Pulling P-Access-Network-Info From SIP Header |
ASTERISK-29395: chan_sip: Pulling P-Access-Network-Info From SIP Header |
ASTERISK-29396: continuous message (unable to authenticate) in asterisk foreground |
ASTERISK-29397: pjsip: Asterisk isn't tolerant of RFC8760 UASs |
ASTERISK-29398: app_voicemail: Realtime voicemail doesn't cache mailbox new/old count at startup |
ASTERISK-29399: app_voicemail: Crash when using app_voicemail with IMAP |
ASTERISK-29400: ***CLI> *** stack smashing detected ***: <unknown> terminated** |
ASTERISK-29401: No AMI event for queue reset stats |
ASTERISK-29402: res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform does not support it |
ASTERISK-29403: media: Early media not relayed to calling leg |
ASTERISK-29404: Consolidate res_pjsip_messaging fixes for domain name |
ASTERISK-29405: Configure your VoIP Asterisk server |
ASTERISK-29406: Asterisk Separate Signaling IP From Media IP |
ASTERISK-29407: chan_local: Filtering audio formats should not occur on removed streams |
ASTERISK-29408: app_queue: Queue events don't occur after caller leaves queue, but when caller hangs up |
ASTERISK-29409: ARI / endpoint subscription : FRACK when removing an endpoint |
ASTERISK-29410: Dialers get error after working fine for some time |
ASTERISK-29411: Crash in pjsip_msg_find_hdr_by_name |
ASTERISK-29412: SIP and/or PJSIP Event Handling Stops |
ASTERISK-29413: chan_pjsip: reply to erroneous NOTIFY not transmitted with 2 transports |
ASTERISK-29414: res_pjsip_outbound_publish reload with config change results in FRACK! error |
ASTERISK-29415: Crash in PJSIP TLS transport |
ASTERISK-29416: Client Hello sent as TLSv1.3 |
ASTERISK-29417: Hangup Handlers reporting incorrect CDR(duration) and CDR(billsec) when added from a Queue Gosub |
ASTERISK-29418: Segmentation fault |
ASTERISK-29419: Testing |
ASTERISK-29420: Once more with feeling |
ASTERISK-29421: Testing |
ASTERISK-29422: CUBE based recording - Media forking on voice Gateway |
ASTERISK-29423: CUBE based recording - Media forking on voice Gateway |
ASTERISK-29424: stasis: Crash on publishing with no topic |
ASTERISK-29425: Testing 123 |
ASTERISK-29426: FAXOPT is giving false failure status |
ASTERISK-29427: Queue in round robin is not ringing agents for the correct amount of time |
ASTERISK-29428: DTMF on progress results in infinite loop if progress followed by hangup received |
ASTERISK-29429: asterisk deadlock issue |
ASTERISK-29430: Cyber security library usage in asterisk |
ASTERISK-29431: Minimum and maximum dialplan functions |
ASTERISK-29432: New function to allow access to any channel |
ASTERISK-29433: res_rtp_asterisk: Server reflexive candidates use incorrect raddr for RTCP |
ASTERISK-29434: Asterisk reveals pjproject version in STUN packets |
ASTERISK-29435: Asterisk crash occurs when reading audio callback in control stream file. |
ASTERISK-29436: core/channel : DTMF emulation doesn't queue the end when last RTP received is the END frame |
ASTERISK-29437: ARI / channel events: events emitted after ChannelDestroyed |
ASTERISK-29438: TURN Server never added to ICE candidate list. |
ASTERISK-29439: func_volume: Volume function can't be read |
ASTERISK-29440: app_confbridge: Allow ConfBridge answer to be suppressed |
ASTERISK-29441: Core reload making TCP endpoints go offline |
ASTERISK-29442: app_dial: Expand A option to allow announcement playback to caller |
ASTERISK-29443: facing issues while installing Asterisk 16.18.0 |
ASTERISK-29444: Add application to wait for condition |
ASTERISK-29445: res_calendar: Crash when checking if calendar is busy |
ASTERISK-29446: app_confbridge: New ConfKick application |
ASTERISK-29447: Is the SIP response code accessible through the AMI |
ASTERISK-29448: In PJSIP how to transfer a call to telephone extension (physical extension) |
ASTERISK-29449: Gruu or oubound in PJSIP |
ASTERISK-29450: Allow setting channel variables using Originate application |
ASTERISK-29451: RTP IP issue, Reverse Proxy using firewalld and OpenVPN |
ASTERISK-29452: Invalid Input Value for enum yesno_values in SQL Query |
ASTERISK-29453: alembic: incoming_call_offer_pref and outgoing_call_offer_pref missing in "ps_endpoints" table |
ASTERISK-29454: New application to reload modules |
ASTERISK-29455: Local channels (dialed using Originate dialplan application) play back gsm files over ulaw files when both exist |
ASTERISK-29456: pjsip: Crash when getting transport based on type after resolution |
ASTERISK-29457: ConfBridge Application is not working in Asterisk |
ASTERISK-29458: Named ACLs and musiconhold is not reloaded by MySQL RealTime. |
ASTERISK-29459: Missing configuration from PJSIP to SIP conversion script |
ASTERISK-29460: Recognize application/hook-flash in PJSIP |
ASTERISK-29461: Contact header format for inbound calls. One way audio inbound calls behind NAT. |
ASTERISK-29462: Hi, I have my sip account and when I implement python code tutorial, it only works with extensions, not when I use extension to make phone call |
ASTERISK-29463: ARI - PlaybackFinish Error status and events |
ASTERISK-29464: ARI - PlaybackFinish skip error events |
ASTERISK-29465: PJSIP/Outreg taskprocessor overload during simple outage |
ASTERISK-29466: pjsip: Asserts (when built in developer mode) when the call hangs up a few seconds before it would send UPDATE |
ASTERISK-29467: Function CURL not installed. |
ASTERISK-29468: Asterisk PHPAGI |
ASTERISK-29469: Transfer target remains on hold after remote attended transfer |
ASTERISK-29470: sip_to_pjsip script improperly combines registration config |
ASTERISK-29471: Realtime queue_members does not respect linear strategy |
ASTERISK-29472: res_pjsip: OLI/ANI2 support missing |
ASTERISK-29473: 18.4 Crash on pjsip_inv_send_msg |
ASTERISK-29474: core: Exceptionally long queue length with Record and ConfBridge |
ASTERISK-29475: SayNumber triggers WARNING if caller hangs up during application execution |
ASTERISK-29476: res_stir_shaken: Blind SSRF vulnerabilities |
ASTERISK-29477: Function to asynchronously store digits dialed |
ASTERISK-29478: Function to drop frames in the TX or RX directions |
ASTERISK-29479: [patch] Channels are not put on hold for Session Progress with inactive audio |
ASTERISK-29480: fixedjitterbuffer contains an un-wrappered assert that triggers on a negative time slew |
ASTERISK-29481: Asterisk Randomly Crash with astobj2.c: FRACK!, Failed assertion bad magic number |
ASTERISK-29482: Wiki page for ARI and AGI bindings not updated for ages |
ASTERISK-29483: BYE is not send from Asterisk to another SIP address after sbc goes down |
ASTERISK-29484: pjsip: Wrong supported header after SIP handshake |
ASTERISK-29485: core: Inband generation of tones for Busy() and Congestion() may not occur |
ASTERISK-29486: Hint-like extension value lookup function without device state |
ASTERISK-29487: ARI : channel creation takes long time using ARI |
ASTERISK-29489: Add mail application |
ASTERISK-29490: Failed User Authentication |
ASTERISK-29491: Failed User Authentication |
ASTERISK-29492: Failed User Authentication |
ASTERISK-29493: app_stack: Add ReturnIf application |
ASTERISK-29494: cdr_adaptive_odbc: Prevent throwing warnings if CDR filtering is used |
ASTERISK-29495: Return integer instead of float if response is a whole number |
ASTERISK-29496: Add SendMF application |
ASTERISK-29497: Add conditional branch applications |
ASTERISK-29498: [1] Got CPG but we didn't send IAM on CIC 5 PC 4470 ÿ[1] ignoring... asterisk outgoing problem,, |
ASTERISK-29499: sip_to_pjsql.py: Fails to write SQL file |
ASTERISK-29500: sound_has_left |
ASTERISK-29501: ARI - Stasis Playback doesn't hangup call when processing a list of invalid files |
ASTERISK-29502: res_pjsip_config_wizard: doesn't take into account setting "identify/match" |
ASTERISK-29503: Updated identify/match syntax not supported by config wizard |
ASTERISK-29504: PJSIP deltree contact still sending qualify packets |
ASTERISK-29505: res_pjsip_registrar: Expires all contacts from realtime table, even if table is shared across multiple servers |
ASTERISK-29506: res_pjsip: Not able to update existing contact entry in realtime |
ASTERISK-29507: STUN timeout is silently delaying calls |
ASTERISK-29508: STUN server address refresh |
ASTERISK-29509: Callerid not being set via AMI Originate |
ASTERISK-29510: New pattern matcher matches invalid patterns |
ASTERISK-29511: Outbound not working from nat |
ASTERISK-29512: bridge_softmix: Not reinviting in all cases with video |
ASTERISK-29513: statsd: Remove non-standard metric type Meter |
ASTERISK-29514: ari: Audiosocket segfault when no data specified |
ASTERISK-29515: app_queue: QueueSummary and QueueStatus events don't exist in documentation |
ASTERISK-29516: app_senddtmf / local: Sending DTMF does not work when not answered |
ASTERISK-29517: logger: JSON: add CEE fields or a generic template for JSON structure |
ASTERISK-29518: sig_analog: FCG_CAMA fails to signal ANI spill when using MF signaling |
ASTERISK-29519: ROC value not incremented in SRTP |
ASTERISK-29520: chan_pjsip: Sometimes no ringback after change of handling 180/SDP in ASTERISK-29105 |
ASTERISK-29521: No ChannelHold and ChannelUnhold events using ARI |
ASTERISK-29522: segfault in pjsip |
ASTERISK-29523: cdr.c: FRACK!, Failed assertion user_data is NULL (0) |
ASTERISK-29524: The patterns "*" and "#" do not work when calling via a PJSIP trunk |
ASTERISK-29525: PJSIP remove_existing unavailable contacts |
ASTERISK-29526: G729 audio gets corrupted by Asterisk due to smoother |
ASTERISK-29527: res_http_media_cache: Cleanup audio format lookup in HTTP requests |
ASTERISK-29528: Add support for multiple files for agent announcements |
ASTERISK-29529: Add custom logging level |
ASTERISK-29530: Test Issue for Gerrit Hooks |
ASTERISK-29531: Add SAYFILES function |
ASTERISK-29532: Issue with maxptime and VolTE (Voice Over LTE) |
ASTERISK-29533: Automatic CANCEL signal from the caller |
ASTERISK-29534: res_rtp_asterisk: T.140 RED sending characters constantly |
ASTERISK-29535: Segmentation fault in libasteriskpj.so.2 |
ASTERISK-29536: AMI error: command blacklisted |
ASTERISK-29537: Early Media not being passed on to bridge |
ASTERISK-29538: Early Media not being passed on to bridge |
ASTERISK-29539: Segmentation fault at ast_writestream() when write handler not defined (happens with OGG/Speex) |
ASTERISK-29540: aelparse: include of context with timings fails |
ASTERISK-29541: app_morsecode: Add American Morse code |
ASTERISK-29542: Add audio scrambler |
ASTERISK-29543: app_originate: Allow specifying codec(s) to use |
ASTERISK-29544: Media Cache - Delayed remote sound file retrieve delays all playbacks |
ASTERISK-29545: Early Media with Multiple Calls |
ASTERISK-29546: Add tone detection module |
ASTERISK-29547: cdr: Wrong CDR(dst) after blind transfer |
ASTERISK-29548: app_meetme: Deprecated in 19, to be removed in 21 |
ASTERISK-29549: app_osploop: Deprecated in 19, to be removed in 21 |
ASTERISK-29550: chan_alsa: Deprecated in 19, to be removed in 21 |
ASTERISK-29551: chan_mgcp: Deprecated in 19, to be removed in 21 |
ASTERISK-29552: chan_skinny: Deprecated in 19, to be removed in 21 |
ASTERISK-29553: res_pktccops: Deprecated in 19, to be removed in 21 |
ASTERISK-29554: cdr_mysql: Deprecated in 1.8, to be removed in 19 |
ASTERISK-29555: app_mysql: Deprecated in 1.8, to be removed in 19 |
ASTERISK-29556: app_getcpeid: Deprecated in 16, to be removed in 19 |
ASTERISK-29557: app_ices: Deprecated in 16, to be removed in 19 |
ASTERISK-29558: app_macro: Deprecated in 16, to be removed in 21 |
ASTERISK-29559: app_fax: Deprecated in 16, to be removed in 19 |
ASTERISK-29560: app_url: Deprecated in 16, to be removed in 19 |
ASTERISK-29561: app_image: Deprecated in 16, to be removed in 19 |
ASTERISK-29562: app_nbscat: Deprecated in 16, to be removed in 19 |
ASTERISK-29563: app_dahdiras: Deprecated in 16, to be removed in 19 |
ASTERISK-29564: cdr_syslog: Deprecated in 16, to be removed in 19 |
ASTERISK-29565: chan_oss: Deprecated in 16, to be removed in 19 |
ASTERISK-29566: chan_phone: Deprecated in 16, to be removed in 19 |
ASTERISK-29567: chan_sip: Deprecated in 17, to be removed in 21 |
ASTERISK-29568: chan_nbs: Deprecated in 16, to be removed in 19 |
ASTERISK-29569: chan_misdn: Deprecated in 16, to be removed in 19 |
ASTERISK-29570: chan_vpb: Deprecated in 16, to be removed in 19 |
ASTERISK-29571: res_config_sqlite: Deprecated in 16, to be removed in 19 |
ASTERISK-29572: res_monitor: Deprecated in 16, to be removed in 21 |
ASTERISK-29573: conf2ael: Deprecated in 16, to be removed in 19 |
ASTERISK-29574: muted: Deprecated in 16, to be removed in 19 |
ASTERISK-29575: app_milliwatt: Milliwatt application doesn't use the proper timings |
ASTERISK-29576: Scope Tracing: format string is not a string literal |
ASTERISK-29577: Scope Tracing: illegal storage on function |
ASTERISK-29578: app_queue: Custom device state using included hints do not update |
ASTERISK-29579: codec_opus: 16000 not WB, 24000 not SWB. |
ASTERISK-29580: codec_opus: Version of included libopus? |
ASTERISK-29581: compiler specific extension makes it impossible to compile asterisk on some platforms |
ASTERISK-29582: res_pjproject: Can't map pjproject log messages to Asterisk TRACE |
ASTERISK-29583: [patch] BuildSystem: User CFLAGS should always have the last say. |
ASTERISK-29584: cdr_mysql: Remove deprecated module |
ASTERISK-29585: app_mysql: Remove deprecated module |
ASTERISK-29586: app_ices: Remove deprecated module |
ASTERISK-29587: app_fax: Remove deprecated module |
ASTERISK-29588: app_url: Remove deprecated module |
ASTERISK-29589: app_image: Remove deprecated module |
ASTERISK-29590: app_nbscat: Remove deprecated module |
ASTERISK-29591: app_dahdiras: Remove deprecated module |
ASTERISK-29592: cdr_syslog: Remove deprecated module |
ASTERISK-29593: chan_oss: Remove deprecated module |
ASTERISK-29594: chan_phone: Remove deprecated module |
ASTERISK-29595: chan_nbs: Remove deprecated module |
ASTERISK-29596: chan_misdn: Remove deprecated module |
ASTERISK-29597: chan_vpb: Remove deprecated module |
ASTERISK-29598: res_config_sqlite: Remove deprecated module |
ASTERISK-29599: conf2ael: Remove deprecated application |
ASTERISK-29600: muted: Remove deprecated application |
ASTERISK-29601: moduleinfo: Add replacement module information |
ASTERISK-29602: res_monitor: Disable building by default. |
ASTERISK-29603: res_pjsip: UPDATE/re-INVITE not sent when "timers=always" is specified in pjsip.conf |
ASTERISK-29604: ari: Segfault with lots of calls |
ASTERISK-29605: chan_iax2: Add ANI2 |
ASTERISK-29606: garbled or absent bluetooth audio due to hardcoded frame size |
ASTERISK-29607: Two incoming calls get hangup in different bridges. |
ASTERISK-29608: asterisk ari client recording stream to aws lex api (PCM) format |
ASTERISK-29609: Subsequent 'ael reload' will cause a lock up |
ASTERISK-29610: ARI channels/move command causes segfault if app name is null |
ASTERISK-29611: how to get AMI logs to asterisk console |
ASTERISK-29612: bridge_basic: Don't throw warning if attended transfer is cancelled |
ASTERISK-29613: chan_sip.c:22332 handle_request_invite: Unable to create/find SIP channel for this INVITE |
ASTERISK-29614: app_agent_pool: XML Doc: unterminated entity reference |
ASTERISK-29615: app_read: Allow reading the digit # |
ASTERISK-29616: res_rtp_asterisk: sqrt(.) requires the header math.h. |
ASTERISK-29617: res_adsi: build although deprecated. |
ASTERISK-29618: ConfBridge errors on creation conference room |
ASTERISK-29619: Unexpected occasional restarts during confbridge calls |
ASTERISK-29620: Changing queue strategy to Linear does not order agents properly |
ASTERISK-29621: Handling of "external_signaling_address" & "external_media_address" |
ASTERISK-29622: ARI: external media create doesn't use body parameter |
ASTERISK-29623: app_queue: Un-ordered answering of callers |
ASTERISK-29624: Contact identifier is not updated when FDQN resolves to a new address |
ASTERISK-29625: srtp cryptos accepted if not enabled |
ASTERISK-29626: app_stack: Include calling location if attempting to branch to nonexistent location |
ASTERISK-29627: Add STRBETWEEN function |
ASTERISK-29628: Add file and directory functions |
ASTERISK-29629: ARI external media channel creation doesn't set option data |
ASTERISK-29630: Asterisk is unable to read extended number format terminfo files |
ASTERISK-29631: pjsip - MessageSend error |
ASTERISK-29632: Add option to Application_VoiceMail to suppress instructions only when a custom greeting is present |
ASTERISK-29633: ast_coredumper: Modules directory sometimes not being detected correctly |
ASTERISK-29634: res_snmp: gcc 11 needs -fPIC to compile correctly |
ASTERISK-29635: MP3Player don' t work with actual mpg123 versions |
ASTERISK-29636: pjsip deadlock |
ASTERISK-29637: Add support for future dates in Say.c |
ASTERISK-29638: res_pjsip_session: No video after early media |
ASTERISK-29639: app_voicemail: Voicemail message playback ignores tz setting |
ASTERISK-29640: Add native dial pulsing |
ASTERISK-29641: Intermittent high peer latency between Media Gateway and Genesys/Opensips Servers |
ASTERISK-29642: Does UDP error causes high peer latency? |
ASTERISK-29643: Outbound calls without Soft Phone |
ASTERISK-29644: pjsip: Can't register to sip if username has "@" in string |
ASTERISK-29653: Failed Authentication on Device (Invite Request) |
ASTERISK-29654: pjproject includes trailing whitespace in sdp format attributes |
ASTERISK-29655: res_pjsip_session: No video to caller if no camera available |
ASTERISK-29656: Add CHANNEL_EXISTS function |
ASTERISK-29657: Asterisk queue stops or hangs |
ASTERISK-29658: app_queue: Multiple members in same queue with same state interface don't all reflect proper state |
ASTERISK-29659: res_pjsip: Authentication fails with wildix |
ASTERISK-29660: Build failure when disabling PJSIP support |
ASTERISK-29661: func_vmcount: Add support for multiple mailboxes |
ASTERISK-29662: Add mix option to Playback application for say and filename |
ASTERISK-29663: messaging: AMI MessageSend does not support same parameters as dialplan application |
ASTERISK-29664: PJSIP processing token with % incorrectly |
ASTERISK-29665: pjproject silently drops registration requests using RFC 8898 syntax |
ASTERISK-29666: There is no video display when the call is in progress. |
ASTERISK-29667: Asterisk sends with the wrong Nonce to the Telecom operator for the Registration Packet. |
ASTERISK-29668: ari: Listing bridges fails when dialing bridge exists |
ASTERISK-29669: Check user response. |
ASTERISK-29670: pjsip is not working correctly |
ASTERISK-29671: res_rtp_asterisk: memory leak |
ASTERISK-29672: Need to send Display Name in From Header usign node-ari-client |
ASTERISK-29673: app_read: Fix null pointer crash regression |
ASTERISK-29674: Adjust for 64bit time_t |
ASTERISK-29675: testsuite: There's no way to copy key files to the proper locations |
ASTERISK-29676: It's hard to navigate the testsuite's asterisk logs |
ASTERISK-29677: Testsuite: Have message_send_ami use destination |
ASTERISK-29678: Asterisk not logging IP addresses associated with device authentication failures |
ASTERISK-29679: Cancel Call forward for SIP one extn to other extn |
ASTERISK-29680: We have a randomly core dump of asterisk |
ASTERISK-29681: chan_sip: Add custom SIP tag parameters |
ASTERISK-29682: Squash compiler issues generated by gcc 11 |
ASTERISK-29683: pjsip: Recalcitrant softphone sends absurd, wrong UDP port in Via: header (REGISTER) |
ASTERISK-29684: configure.ac doesn't handle the check for OPENSSL_BIO_METHOD correctly |
ASTERISK-29685: pbx_ael: Infinite loop on reload |
ASTERISK-29686: Module loading should be more reasonable |
ASTERISK-29687: Figure out the future of format_mp3 'cuz it won't compile anymore |
ASTERISK-29688: Cryptograph DTMF in Asterisk Log and Asterisk CLI in Version 11.6 or higher |
ASTERISK-29689: Cryptograph DTMF in Asterisk Log and Asterisk CLI in Version 11.6 or higher |
ASTERISK-29690: Cryptograph DTMF in Asterisk Log and Asterisk CLI in Version 11.6 or higher |
ASTERISK-29691: stun: Not all users provide a dst to ast_stun_request |
ASTERISK-29692: Asterisk stops responding to SIP OPTIONS requests from other SIP servers or asterisk servers via LAN or WAN with PJSIP module |
ASTERISK-29693: Using --with-crypto and --with-ssl fails on a recompile |
ASTERISK-29694: Is ACN works? |
ASTERISK-29695: SAY.CONF wrong logic when converting 24hour time to say 12 hour am/pm |
ASTERISK-29696: say.conf: wrong logic when converting 24 hour time to 12 hour am/pm |
ASTERISK-29697: Remove experimental 'say' application |
ASTERISK-29698: Segfault if sorcery object_lifetime_maximum and qualify_frequency the same value |
ASTERISK-29699: ConfBridge doesn't detect disconnect from webrtc |
ASTERISK-29700: DTMF is sent in RTP events even if remote 200 OK omits telephone-event |
ASTERISK-29701: Add assertion application |
ASTERISK-29702: sig_analog: Fix truncated buffer copy |
ASTERISK-29703: res_pjsip_callerid: Fix OLI parsing |
ASTERISK-29704: res_pjsip_pubsub: Treats SUBSCRIBE with Expires 0 as invalid when it is valid |
ASTERISK-29705: app_read: Fix custom terminator functionality regression |
ASTERISK-29706: func_json: Add JSON parsing function |
ASTERISK-29707: chan_iax2: Allow both key and secret to be specified at dial time |
ASTERISK-29708: format_mp3: Clang 13 warns about the unused but set variable clip. |
ASTERISK-29709: res_snmp: Not build on recent Debian distributions. |
ASTERISK-29710: stasis: Clang 13 warns about the unused but set variable dispatched. |
ASTERISK-29711: aelparse: GCC 11.2 found two maybe uninitialized |
ASTERISK-29712: format_mp3: GCC 11.2 found an outside array bounds |
ASTERISK-29713: GCC 11.2: two stringop-overread |
ASTERISK-29714: Spelling errors |
ASTERISK-29715: app_voicemail: Refactor email generation functions |
ASTERISK-29716: Dtmf Problem With Payload 96 |
ASTERISK-29717: res_config_sqlite: not removed in makeopts.in |
ASTERISK-29719: I'm unable to compile Asterisk 18.8.0 |
ASTERISK-29720: res_tonedetect: Add call progress tone detection |
ASTERISK-29721: channel.c: FRACK!, Failed assertion user_data is NULL |
ASTERISK-29722: test_timezone_watch breaks during DST to ST transition |
ASTERISK-29723: Subsystem alert cleared unexpectedly at 4 [test_taskprocessor.c:subsystem_alert:290]: Global alert cleared unexpectedly at 4 |
ASTERISK-29724: BuildSystem: In POSIX sh, == in place of = is undefined. |
ASTERISK-29725: COMPILE_DOUBLE exposes issues in logger.c and channel.c with GCC 11.2 |
ASTERISK-29726: Add Asterisk External Application Protocol (AEAP) implementation |
ASTERISK-29727: Add type for JSON stasis message RTCP Report Received/Sent |
ASTERISK-29728: menuselect: Disabled by default modules that are enabled are always recompiled |
ASTERISK-29729: Incompatibility with newer spandsp releases (3.0.0+) |
ASTERISK-29730: Segfault in __ao2_ref if refdebug = yes |
ASTERISK-29731: res_pjsip_outbound_publish: Reload results in invalid state and usage of freed memory |
ASTERISK-29732: progdocs: Fix grouping for latest Doxygen |
ASTERISK-29733: progdocs: Avoid name with Doxygen \file |
ASTERISK-29734: progdocs: Use Doxygen \example correctly |
ASTERISK-29735: progdocs: Avoid multiple use of section labels |
ASTERISK-29736: bridge_channel: Fix for Doxygen |
ASTERISK-29737: chan_iax2: Fix for Doxygen |
ASTERISK-29738: when called is busy |
ASTERISK-29739: Add compiler options for FreeBSD |
ASTERISK-29740: apps: Fix for Doxygen |
ASTERISK-29741: tests: Fix for Doxygen |
ASTERISK-29742: addons: Fix for Doxygen. |
ASTERISK-29743: bridges: Fix for Doxygen |
ASTERISK-29744: app_morsecode: Fix deadlock |
ASTERISK-29745: pbx: Add public API for more elegant variable substitution with extensions |
ASTERISK-29746: tcptls.c: TCP client connect fails due to interrupt |
ASTERISK-29747: res_pjsip: Fix for Doxygen |
ASTERISK-29748: bridging: Infinite loop when both Local channel halves in same bridge |
ASTERISK-29749: res_xmpp: Fix for Doxygen |
ASTERISK-29750: stasis: Fix for Doxygen |
ASTERISK-29751: channel: Fix for Doxygen |
ASTERISK-29752: app: Fix for Doxygen |
ASTERISK-29753: parking: Fix for Doxygen |
ASTERISK-29754: odbc: Fix for Doxygen |
ASTERISK-29755: frame: Fix for Doxygen |
ASTERISK-29756: res_ari: Fix for Doxygen |
ASTERISK-29757: TCP: SIP Packets Being Combined |
ASTERISK-29758: configs: Minor updates to sample configs |
ASTERISK-29759: app_sendtext: Add ReceiveText application |
ASTERISK-29760: app_chanspy: Barge not supported when channel in a ConfBridge |
ASTERISK-29761: res: Fix for Doxygen |
ASTERISK-29762: channels: Fix for Doxygen |
ASTERISK-29763: main: Fix for Doxygen |
ASTERISK-29764: chan_misdn: Fix for Doxygen |
ASTERISK-29765: xmldoc: Fix for Doxygen |
ASTERISK-29766: pbx_variables: MSet truncates sets after 24 variables |
ASTERISK-29767: sip.conf |
ASTERISK-29768: How to get Call-ID in this format "1132932127b2d80b7e96a31416373e4@domain.com" from asterisk |
ASTERISK-29769: AMI call flow incorrect linkedId for transferred extension after AttendedTransfer event |
ASTERISK-29770: call disconnects after 15 minutes |
ASTERISK-29771: Crash occurs when 2 realtime sippeers mysql connections are configured and we have a schema warning |
ASTERISK-29772: chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause a coredump |
ASTERISK-29773: progdocs: doxyref.h outdated |
ASTERISK-29774: Deadlock where hints lock blocks contexts lock |
ASTERISK-29775: CPU spike |
ASTERISK-29776: stir/shaken: Requires GNU designator |
ASTERISK-29777: documentation: Standardize example syntax |
ASTERISK-29778: Unable to locate the FreePBX BMO Class 'Get_sysadmin_extensions_limit' |
ASTERISK-29779: progdocs: Hidden code sections with syntax errors. |
ASTERISK-29780: bridge: Sending answer towards called party causes bidirectional audio to drop permanently due to AST_CONTROL_SRCCHANGE |
ASTERISK-29781: My asterisk server crashing repeatedly |
ASTERISK-29782: app_queue: Queue members showing Unavailable for no reason |
ASTERISK-29783: Update documents referencing chan_sip |
ASTERISK-29784: How can I reduce the time taken by wav file to save in /tmp folder |
ASTERISK-29785: res_pjsip_sdp_rtp: Warns on every offered crypto suite |
ASTERISK-29786: cdr.c: FRACK! causing crashes |
ASTERISK-29787: Dependency not met for compilation |
ASTERISK-29788: symbol lookup and trap invalid opcode errors with migrated VM |
ASTERISK-29789: chan_pjsip: Add ADSI support |
ASTERISK-29790: xmldoc: Dump invalid to XML DTD: XSLT |
ASTERISK-29791: xmldoc: Dump invalid to XML DTD: ACO Matchfield |
ASTERISK-29792: xmldoc: Dump invalid to XML DTD: XInclude |
ASTERISK-29793: adsi: CAS is malformed |
ASTERISK-29794: ast_coredumper does not delete results when requested and a specific output dir is set |
ASTERISK-29795: DIALEDPEERNUMBER not set on destination channel for Queue calls |
ASTERISK-29796: app_queue: Invalid agents status on queues. |
ASTERISK-29797: Support for Danish language syntax in VM |
ASTERISK-29798: the call cut between two extensions |
ASTERISK-29799: the call cut between two extensions |
ASTERISK-29800: strings: Fix misusage in comment examples |
ASTERISK-29801: app.c: Throw warnings for nonexistent options |
ASTERISK-29802: app_sf: Add full tech-agnostic SF support |
ASTERISK-29803: pbx_variables: cp4 variables is used uninitialized |
ASTERISK-29804: bundled_pjproject: sip_inv is missing multipart support in some cases |
ASTERISK-29805: asterisk AMI action to do confference |
ASTERISK-29806: app_queue: extension state incorrect |
ASTERISK-29807: cli: add module refresh command |
ASTERISK-29808: cdr: allow disabling CDR by default |
ASTERISK-29809: curl, stir_shaken: refactor curl code |
ASTERISK-29810: app_signal: Add channel signaling applications |
ASTERISK-29811: func_logic: IF func (acf_if) inconsistent behavior |
ASTERISK-29812: GotoIF Loop prevention |
ASTERISK-29813: res_pjsip_session doesn't support multipart message bodies |
ASTERISK-29814: Very High CPU uses only 80 con-currents calls on asterisk via AMI |
ASTERISK-29815: dsp: Define magic number as macro |
ASTERISK-29816: SAY_DTMF_INTERRUPT channel variable is not honored |
ASTERISK-29817: gethostbyname_r is misdetected on NetBSD and causes a build failure |
ASTERISK-29818: Build failure on NetBSD due to hmac function collision |
ASTERISK-29819: utils.c: Remove all usages of ast_gethostbyname() |
ASTERISK-29820: cli: Add command to evaluate a function |
ASTERISK-29821: Deadlock in bridge_channel_internal_join() on local channels. |
ASTERISK-29822: cli: Typing \? freezes the CLI permanently with remote console |
ASTERISK-29823: how to transfer a call into three or more sip extensions in vxml script |
ASTERISK-29824: It's hard to make changes to bundled pjproject |
ASTERISK-29825: func_frameintercept: add function to intercept control frames |
ASTERISK-29826: asterisk AMI action to do confference |
ASTERISK-29827: Support for Nordic language syntax in Queues |
ASTERISK-29828: pbx: Add mechanism for applications to preemptively validate dialplan |
ASTERISK-29829: app_mp3: Throw warning if attempting to play a nonexistent stream |
ASTERISK-29830: ami: Add AMI event for Wink |
ASTERISK-29831: Queue don't play "thank-you" when here is no hold time announcements |
ASTERISK-29832: Enable pickup on channel after having received 183 Progress |
ASTERISK-29833: asterisk lot of open fd |
ASTERISK-29834: pjsip: Incorrect IP in contact header when using PJSIP TLS with dynamic IP addresses (intermittently) |
ASTERISK-29838: ${SQL_ESC()} not correctly escaping a terminating \ |
ASTERISK-29839: res_pjsip: Failover occurs after CANCEL |
ASTERISK-29840: func_channel: Add LASTCONTEXT and LASTEXTEN fields |
ASTERISK-29841: APP Queue Deadlock |
ASTERISK-29842: Do not change 180 Ringing to 183 Progress even if early_media already enabled |
ASTERISK-29843: Session timers get removed on UPDATE |
ASTERISK-29844: res_pjsip_logger: Logger output can show multiple packets on connection oriented transports |
ASTERISK-29845: res_pjsip_outbound_registration: Show time remaining until registration lapses |
ASTERISK-29846: channels: bad ao2 ref causes crash |
ASTERISK-29847: pbx_variables: ASTSBINDIR is missing |
ASTERISK-29848: documentation: Document special system and channel variables |
ASTERISK-29849: pbx_variables: Add variable registration and validation |
ASTERISK-29850: ast_get_tid() not implemented for NetBSD |
ASTERISK-29851: rdtsc is not enabled (stubbed out) on NetBSD |
ASTERISK-29852: make_version uses GNU-ism that break git-svn-id parsing on NetBSD |
ASTERISK-29853: ami: Allow events to be globally disabled |
ASTERISK-29854: func_frame_drop: fix buffer usage typo |
ASTERISK-29855: frame.h: fix CNG documentation typo |
ASTERISK-29856: res_rtp_asterisk: Invalid comparison creates unreachable code |
ASTERISK-29857: res_tonedetect: fix logic errors in code |
ASTERISK-29858: Regression: Using external pjproject not working after "hack" commit |
ASTERISK-29859: VoiceMailMain() fails when encountering non-numeric CALLERID(num) |
ASTERISK-29860: Segfault in Asterisk 18.8.0 |
ASTERISK-29861: asterisk.h: add macro for curl user agent |
ASTERISK-29862: dsp: non-PCM/SLIN codecs are rejected |
ASTERISK-29863: codecs: clicking is introduced onto channels |
ASTERISK-29864: Bridge() does not work properly when used on a channel in a ConfBridge |
ASTERISK-29865: ChannelRedirect() does not work properly when used on a channel in a ConfBridge |
ASTERISK-29866: cli: add core dump information to core show settings |
ASTERISK-29867: configure fails if libsrtp dev files are not installed |
ASTERISK-29868: how does astrick dails out |
ASTERISK-29869: rtp sequence number can skip after DTMF under certain bridges |
ASTERISK-29870: chan_sip doesn't send NOTIFY if mailbox not fully specified in sip.conf |
ASTERISK-29871: res_prometheus: Failure to load causes FRACKs |
ASTERISK-29872: res_stir_shaken: Resource exhaustion with large files |
ASTERISK-29873: [patch] Queue Realtime load |
ASTERISK-29874: testsuite notify_after_register/realtime sipp invalid sipp scenarios |
ASTERISK-29875: queue_member_forward - failure if refer comes after initial hangup |
ASTERISK-29876: app_queue: Add music on hold option |
ASTERISK-29877: app_mf: Allow reading a maximum number of digits |
ASTERISK-29878: chan_dahdi: memory leak in dahdi_restart |
ASTERISK-29879: res_musiconhold: Music on hold restarts after positon announcement |
ASTERISK-29880: PJSIP queues reach limits |
ASTERISK-29881: Segmentation fault when using PJSIP + TURN server + WebRTC |
ASTERISK-29882: Occasional segfaults in production |
ASTERISK-29883: Asterisk 16 GSM gateway |
ASTERISK-29884: logger.h: logger macros cause install to fail |
ASTERISK-29885: How to record a video call in Asterisk? |
ASTERISK-29886: Asterisk AMI sends not-valid XML |
ASTERISK-29887: app_confbridge: CONFBRIDGE template modifications are wrongly discarded in certain bridge joins |
ASTERISK-29888: res_pjsip_outbound_authenticator_digest: ABRT attempting to clean up auth_sess |
ASTERISK-29889: asterisk.conf transmit_silence does not work in VoiceMail() |
ASTERISK-29890: chan_dahdi: Cut through audio immediately on selected channels |
ASTERISK-29891: [patch] provide a display name for RLS subscriptions |
ASTERISK-29892: Audio Streaming |
ASTERISK-29893: deadlock during bridge |
ASTERISK-29894: Wrong grammar in German recording of 'vm-Old' |
ASTERISK-29895: chan_iax2: Fix misaligned spacing in iax2 show netstats printout |
ASTERISK-29896: xmldocs: Add since tag |
ASTERISK-29897: channels: Increase core debug levels for chatty debugs |
ASTERISK-29898: documentation: Add default attributes to documentation |
ASTERISK-29899: features: Add advanced transfer initiation options |
ASTERISK-29900: app_mp3: Document and warn about https incompatibility |
ASTERISK-29901: Call to Listen Live Radio / Audio Stream |
ASTERISK-29902: no color for the cli |
ASTERISK-29903: Asterisk Software Bill of Materials (SBOM) |
ASTERISK-29904: RLS: Batched Notifications stop working |
ASTERISK-29905: OSX: bininstall launchd issue on cross-platfrom build |
ASTERISK-29906: [patch] update RLS to reflect the changes to the lists |
ASTERISK-29907: res_pjsip, app_confbridge: Video call through ConfBridge with normal endpoints causes infinite loop/crash |
ASTERISK-29908: pjsip: Asterisk asserts on nonexistent registration auth and crashes |
ASTERISK-29909: app_queue: Add support for withdrawing a call |
ASTERISK-29910: chan_iax2: Fixed jitterbuffer does not work with IAX2 channels |
ASTERISK-29911: app_userevent: Open parentheses or bracket in body breaks UserEvent app |
ASTERISK-29912: res_pjsip: module reload disables logging |
ASTERISK-29913: func_json: Adds multi-level and array parsing to JSON_DECODE |
ASTERISK-29914: 16.24.0, 18.10.0 failed to compile |
ASTERISK-29915: 18.10.0 failed to compile |
ASTERISK-29916: ast_fileexists filename is empty or NULL |
ASTERISK-29917: ami: FilterList action doesn't exist |
ASTERISK-29918: res_pjsip_refer: Can not get custom headers from REFER |
ASTERISK-29919: CRASH ASTERISK |
ASTERISK-29920: app_voicemail: Warn if trying to manage nonexistent mailbox |
ASTERISK-29921: Macro wiki documentation uses Gosub |
ASTERISK-29922: Without Username and Password how to register vodafone sip provider sip |
ASTERISK-29923: docs, LICENSE: pbx.digium.com no longer exists |
ASTERISK-29924: res_config_pgsql: omit "unsupported column type 'text'" error |
ASTERISK-29925: func_db: Warn about malformed key names |
ASTERISK-29926: func_channel: Add TECH_EXISTS |
ASTERISK-29927: Make (compilation) of current Asterisk 18 fails on Ubuntu 20.04 VPS |
ASTERISK-29928: logging messages truncated when using MUSL runtime |
ASTERISK-29929: res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP activity in one way sessions |
ASTERISK-29930: Webphone - webRTC |
ASTERISK-29931: Option to allow a user to not hear the join sound on enter but everyone else can |
ASTERISK-29932: res_pjsip_sdp_rtp.c::create_rtp() binds to UDP6 for UDP4 peer on BSD |
ASTERISK-29933: cdr_adaptive_odbc: datetime type not supported |
ASTERISK-29934: func_channel: Invalid memory management in CHANNELS can cause a crash |
ASTERISK-29935: build: Remove leftover build references |
ASTERISK-29936: app_confbridge / translate: unable to build translation path |
ASTERISK-29937: res_rtp_asterisk: Resending of NACK requested packet sends different packet |
ASTERISK-29938: res_pjsip: Restoration of rewritten Contact on egress does not restore transport |
ASTERISK-29939: agi: Fix xmldoc bug with set music |
ASTERISK-29940: general: Add since tags to xmldocs |
ASTERISK-29941: chan_pjsip: Add ability to send flash events |
ASTERISK-29942: Long time latency time with SIPML5 et Asterisk server |
ASTERISK-29943: file.c: seeking to negative file offset is not prevented |
ASTERISK-29944: Bundled PJSIP 2.12 |
ASTERISK-29945: pjproject: Security fixes for things |
ASTERISK-29946: testsuite: timeout can't be overridden to higher values through argument |
ASTERISK-29947: testsuite: Premature shutdown causes tests to fail |
ASTERISK-29948: iostream: Infinite TCP timeout writing data |
ASTERISK-29949: chan_dahdi.c:13797 dahdi_ss7_error |
ASTERISK-29950: SayNumber can handle '01' to '07', but not '08' or '09' |
ASTERISK-29951: app_mf, app_sf: Return -1 on hangup |
ASTERISK-29952: how to set a callerid in originate command in php script |
ASTERISK-29953: Asterisk sends 488 Unacceptable when the Bind port of RTCP fails. |
ASTERISK-29954: app_meetme: Emit warning if conference not found |
ASTERISK-29955: chan_sip: SIP route header is missing on UPDATE |
ASTERISK-29956: chan_sip: settings not set correctly - useragent, sdpsession,sdpowner |
ASTERISK-29957: res_pjsip: Transport autoselection is broken on FreeBSD |
ASTERISK-29958: app_dial: Hangup cause disappears if progress received |
ASTERISK-29959: No timestamp on voicemail |
ASTERISK-29960: ari: Retrieving stored recording can returns wrong file |
ASTERISK-29961: RLS: domain part of 'uri' list attribute mismatch with SUBSCRIBE request |
ASTERISK-29962: Study purpose |
ASTERISK-29963: res_rtp_asterisk: mapping->ssrc_invalid on unidirectional videostream after confbridge reinvite |
ASTERISK-29964: Faulty/incomplete sample code in Blog "Building a Channel Driver Part 1" |
ASTERISK-29965: res_pjsip_outbound_registration: Make max registration delay configurable |
ASTERISK-29966: pbx_variables: ast_str_strlen can be wrong |
ASTERISK-29967: pbx_builtins: Add missing documentation |
ASTERISK-29968: func_db: Add a function to return cardinality of keys at prefix |
ASTERISK-29969: webrtc |
ASTERISK-29970: Use pkg-config to find libxml2 headers and libraries |
ASTERISK-29971: Calls not connected while using min_/max_penalty and weight |
ASTERISK-29972: Duplicated ARI events |
ASTERISK-29973: GET PJSIP HEADER IN THE BYE OF A MANUAL DIAL |
ASTERISK-29974: Phone registration issue in Asterisk 18.10.1. |
ASTERISK-29975: Phone registration issue in Asterisk 18.10.1. |
ASTERISK-29976: Should Readme include information about install_prereq script? |
ASTERISK-29977: Asterisk do not respect Agents Priority within Queue` |
ASTERISK-29978: chan_sip/res_rtp: Asterisk do not use the first media format in reply with SDP |
ASTERISK-29979: app_chanisavail: Crash when requesting PJSIP channel |
ASTERISK-29980: build: External binary modules don't use https |
ASTERISK-29981: res_calendar: Asterisk crashes when starting, and will not run |
ASTERISK-29982: Need Help to Install 13.8.0 asterisk installation in ubuntu 14.04 |
ASTERISK-29983: Executing php in asterisk server |
ASTERISK-29984: Crash in stasis_message_dtor on ast_json_free |
ASTERISK-29986: build: Asterisk 18.11.0 doesn't compile when wget isn't available |
ASTERISK-29987: Integer Overflow in Asterisk Scheduler |
ASTERISK-29988: REGRESSION: The build process is requiring xmllint or xmlstarlet ro be installed when it shouldn't |
ASTERISK-29989: app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy |
ASTERISK-29990: chan_dahdi: adding ring cadences is not idempotent on dahdi restart |
ASTERISK-29991: chan_dahdi, callerid: Caller ID does not honor presentation |
ASTERISK-29992: chan_dahdi: Allow pulse and tone dialing to be disabled |
ASTERISK-29993: chan_dahdi: Operator control option borks both lines involved on callee disconnect |
ASTERISK-29994: chan_dahdi: Round robin array size is too small for max number of groups |
ASTERISK-29995: websocket connection could not be accepted |
ASTERISK-29996: tcptls.c:179 handle_tcptls_connection: Unable to set up ssl connection with peer |
ASTERISK-29997: Audio Related Issues with WebRTC over Kamailio integration with Asterisk |
ASTERISK-29998: sla: deadlock when calling SLAStation application |
ASTERISK-29999: pjsip: Get information from 200 OK INVITE reply headers |